Method and system for wireless hearing assistance

ABSTRACT

A method for providing hearing assistance to a user, comprising capturing audio signals by an internal microphone arrangement and supplying the captured audio signals a central signal processing unit; estimating whether a certain type of external audio signal supply device is connected to the audio signal processing unit in order to supply external audio signals to the central signal processing unit, and selecting an audio signal processing scheme according to the estimated type of external audio signal supply device; processing, the captured audio signals and the external audio signals according to the selected audio signal processing scheme; transmitting the processed audio signals to stimulating means worn at or in at least one of the user&#39;s ears via a wireless audio link; and stimulating the user&#39;s hearing by said stimulating means according to the processed audio signals.

BACKGROUND OF THE INVENTION

1. Field of the Invention

The present invention relates to a system for providing hearingassistance to a user, comprising a microphone arrangement for capturingaudio signals, an central signal processing unit for processing thecaptured audio signals, and means for transmitting the processed audiosignals via a wireless audio link to means worn at or in at least one ofthe user's ears for stimulating the hearing of the user according to theprocessed audio signals.

2. Description of Related Art

Usually in such systems the wireless audio link is an FM radio link. Thebenefit of such systems is that sound captured by a remote microphone atthe transmission unit can be presented at a high sound pressure leveland good signal-to-noise ratio (SNR) to the hearing of the user wearingthe receiver unit at his ear(s).

According to one typical application of such wireless audio systems, thestimulating means is a loudspeaker which is part of a receiver unit oris connected thereto. Such systems are particularly helpful for beingused in teaching e.g. (a) normal-hearing children suffering fromauditory processing disorders (APD), (b) children suffering a unilateralloss (one deteriorated ear), or (c) children with a mild hearing loss,wherein the teacher's voice is captured by the microphone of thetransmission unit, and the corresponding audio signals are transmittedto and reproduced by the receiver unit worn by the child, so that theteacher's voice can be heard by the child at an enhanced level, inparticular with respect to the background noise level prevailing in theclassroom. It is well known that presentation of the teacher's voice atsuch enhanced level supports the child in listening to the teacher.

According to another typical application of wireless audio systems thereceiver unit is connected to or integrated into a hearing instrument,such as a hearing aid. The benefit of such systems is that themicrophone of the hearing instrument can be supplemented with orreplaced by the remote microphone which produces audio signals which aretransmitted wirelessly to the FM receiver and thus to the hearinginstrument. FM systems have been standard equipment for children withhearing loss (wearing hearing aids) and deaf children (implanted with acochlear implant) in educational settings for many years.

Hearing impaired adults are also increasingly using FM systems. Theytypically use a sophisticated transmitter which can (a) be pointed tothe audio-source of interest (during e.g. cocktail parties), (b) put ona table (e.g. in a restaurant or a business meeting), or (c) put aroundthe neck of a partner/speaker and receivers that are connected to orintegrated into the hearing aids. Some transmitters even have anintegrated Bluetooth module giving the hearing impaired adult thepossibility to connect wirelessly with devices such as cell phones,laptops etc.

The merit of wireless audio systems lies in the fact that a microphoneplaced a few inches from the mouth of a person speaking receives speechat a much higher level than one placed several feet away. This increasein speech level corresponds to an increase in signal-to-noise ratio(SNR) due to the direct wireless connection to the listener'samplification system. The resulting improvements of signal level and SNRin the listener's ear are recognized as the primary benefits of FM radiosystems, as hearing-impaired individuals are at a significantdisadvantage when processing signals with a poor acoustical SNR.

CA 2 422 449 A2 relates to a communication system comprising an FMreceiver for a hearing aid, wherein audio signals may be transmittedfrom a plurality of transmitters via an analog FM audio link.

Usually the remote wireless microphone of a wireless hearing assistancesystem is a portable or hand-held device which may be used in multipleenvironments and conditions: (a) the remote microphone may be held bythe hearing-impaired person and pointed towards the desired audiosource, such as in a one-to-one conversation to the interlocutor; (b)the remote microphone may be worn around the neck; (c) the remotemicrophone may be put on a table in a conference or restaurantsituation; (d) an external microphone may be connected to the system,which may be worn, for example, in the manner of a lapel microphone or aboom microphone; (e) an external audio source, such as a music player,may be connected to the system.

Usually, the audio signal processing schemes implemented in suchwireless systems are a compromise between all wearing modes andoperation options. Typically, these signal processing schemes, inparticular, the gain model, are fixed, apart from the user's possibilityto manually choose between a few beam forming and noise cancelingoptions, which are commonly referred to as different “zoom” positions.

For hearing instruments it is known to perform an analysis of thepresent acoustic environment (“classifier”) based on the audio signalscaptured by the internal microphone of the hearing instrument in orderto select the most appropriate audio signal processing scheme, inparticular with regard to the compression characteristics, for the audiosignal processing within the hearing instrument based on the result ofthe acoustic environment analysis. Examples of classifier approaches arefound in US 2002/0090098 A1, US 2007/0140512 A1, EP 1 326 478 A2 and EP1 691 576 A2.

In EP 1 691 574 A2 and EP 1 819 195 A2 wireless hearing assistancesystems are described, comprising a transmission unit including abeamformer microphone arrangement and a hearing instrument, wherein aclassifier for analyzing the acoustic environment is located in thetransmission unit and wherein the result provided by the classifier isused to adjust the gain applied to the audio signals captured by thebeam former microphone arrangement in the transmission unit and/or inthe receiver unit/hearing instrument.

EP 1 083 769 A1 relates to a hearing aid system comprising a sensor forcapturing the movements of the user's body, such as an accelerationsensor, wherein the information provided by such sensor is used in aspeech recognition process applied to audio signals captured by themicrophone of the hearing aid.

EP 0 567 535 B1 relates to a hearing aid comprising an accelerometer forcapturing mechanical vibrations of the hearing aid housing in order tosubtract the accelerometer signal from the audio signals captured by theinternal microphone of the hearing aid.

WO 2007/082579 A2 relates to a hearing protection system comprising twoearplugs, which each comprise a microphone and a loudspeaker connectedby wires to a common central audio signal processing unit worn around atthe user's body. A detector is provided for detecting whether externalaudio signals are provided to the central audio signal processing unitfrom an external communication device connected to the central audiosignal processing unit. The output signal of the detector is used toselect an audio signal processing mode of the central audio signalprocessing unit.

US 2004/0136522 A1 relates to a hearing protection system comprising twohearing protection headphones which both comprise anactive-noise-reduction unit. The headphones also comprise a loudspeakerfor reproducing external audio signals supplied from externalcommunication devices. The system also comprises a boom microphone. Adevice detector is provided for controlling the supply of power to theboom microphone depending on whether a external communication device isconnected to the system.

US 2002/0106094 A1 relates to a hearing aid comprising in internal and awireless external microphone. A connection detection circuit is providedfor activating the power supply of the external microphone once theexternal microphone is electrically separated from the hearing aid.

It is an object of the invention to provide for a method for providinghearing assistance using a wireless microphone arrangement, wherein thelistening comfort, such as the signal to noise ratio (SNR), should beoptimized at any time. It is a further object of the invention toprovide for a corresponding wireless hearing assistance system.

SUMMARY OF THE INVENTION

According to the invention, these objects are obtained by a method asdefined in claim 1 and by a system as defined in claim 19, respectively.

The invention is beneficial in that, by estimating whether a certaintype of external audio signal supply device is connected to the centralsignal processing unit and selecting an audio signal processing schemeaccording to the estimated type of external audio signal supply device,the processing of the audio signals captured by the microphonearrangement can be automatically adjusted to the present use situationof the system.

Preferred embodiments of the invention are defined in the dependentclaims.

These and further objects, features and advantages of the presentinvention will become apparent from the following description when takenin connection with the accompanying drawings which, for purposes ofillustration only, show several embodiments in accordance with thepresent invention.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a block diagram of one embodiment of a hearing assistancesystem according to the invention;

FIG. 2 is a block diagram showing in a schematic manner the internalstructure of the central signal processing unit of the system of FIG. 1;

FIG. 3 is an example of a default setting of the output signal level(top) and the corresponding gain (bottom) as a function of the inputsignal level;

FIG. 4 shows examples of deviations from the default setting of FIG. 3for different use modes of a hearing assistance system according to theinvention; and

FIG. 5 shows an example of the gain as a function of the audio signalfrequency for a default setting and for specific use modes of a hearingassistance system according to the invention.

DETAILED DESCRIPTION OF THE INVENTION

FIG. 1 shows a block diagram of an example of a wireless hearingassistance system comprising a transmission unit 10 and at least one earunit 12 which is to be worn at or in one of the user's ears (an ear unit12 may be provided only for one of the two ears of the user, or an earunit 12 may be provided for each of the ears). According to FIG. 1 theear unit 12 comprises a receiver unit 14, which may supply its outputsignal to a hearing instrument 16 which is mechanically and electricallyconnected to the receiver unit 14, for example, via a standardizedinterface 17 (such as a so-called “audio shoe”), or, according to avariant, to a loudspeaker 18, which is worn at least in part in theuser's ear canal (for example, the loudspeaker itself may be located inthe ear canal or a sound tube may extend from the loudspeaker located atthe ear into the ear canal).

The hearing instrument 16 usually will be a hearing aid, such as of theBTE (Behind The Ear)-type, the ITE (In The Ear)-type or the CIC(Completely In the Canal)-type. Typically, the hearing instrument 16comprises one or more microphones 20, a central unit 22 for performingaudio signal processing and for controlling the hearing instrument 16, apower amplifier 24 and a loudspeaker 26.

The transmission unit 10 comprises a transmitter 30 and an antenna 32for transmitting audio signals processed in a central signal processingunit 28 via a wireless link 34 to the receiver unit 14, which comprisesan antenna 36, a receiver 38 and a signal processing unit 40 forreceiving the audio signals transmitted via the link 34 in order tosupply them to the hearing instrument 16 or the speaker 18. The wirelessaudio link 34 preferably is an FM (frequency modulation) link.

Rather than consisting of a receiver unit 14 connected to a hearinginstrument 16 the ear unit 12, as an alternative, may consist of ahearing instrument 16′ into which the functionality of the receiver unit14, i.e. the antenna 36 and the receiver 38, is integrated. Such analternative is also schematically shown in FIG. 1.

The transmission unit 10 comprises a microphone arrangement 42, whichusually comprises at least two spaced-apart microphones M1 and M2, anaudio input 44 for connecting an external audio source 46, e.g. a musicplayer, or an external microphone 48 to the transmission unit 10, adistance sensor 50, an acceleration sensor 52 and an orientation sensor54. In addition, the transmission unit 10 may comprise a second audioinput 44′, so that, for example, the external audio source 46 and theexternal microphone 48 my be connected at the same time to thetransmission unit 10. The transmission unit 10 also may comprise anauxiliary microphone 56 in close mechanical and acoustical contact withthe housing of the transmission unit 10 for capturing audio signalsrepresentative of body noise and/or housing noise. The externalmicrophone 48 may comprise one or several capsules, the signals of whichare further processed in the central signal processing unit 28. Thetransmission unit 10 also comprises a unit 66 which is capable ofdetermining whether and which type of an external audio signal source 46is connected to the audio input 44 and to estimate the type of aexternal microphone 48, when connected to the audio input 44, by sensingat least one electrical parameter, such as the impedance of the externalmicrophone 48.

The transmission unit 10 is designed as a portable unit which may serveseveral purposes: (a) it may be used in a “conference mode”, in which itis placed stationary on a table; (b) it may be used in a “hand-heldmode”, in which it is held in the hand of the user of the ear unit 12;(c) it may be worn around a person's neck, usually a person speaking tothe user of the ear unit 12, such as the teacher in a classroom teachinghearing-impaired persons, or a guide in a museum, etc. (“neck mode”);(d) it may be worn at the body of the user of the ear unit 12, with anexternal microphone 48 and/or an external audio source 46 beingconnected to the transmission unit 10 (“external audio mode”); theexternal audio source 46 may be e.g. a TV set or any kind of audioplayer (e.g. MP3). The transmission unit 10 may in this case also beplaced next to the audio equipment.

FIG. 2 is a block diagram showing in a schematic manner the internalstructure of the central signal processing unit 28 of the transmissionunit 10, which comprises a beam former 58, a classification unit 60including a voice activity detector (VAD), an audio signal mixing/addingunit 62 and an audio signal processing unit 64. The audio signalprocessing unit 64 usually will include elements like a gain model,noise canceling algorithms and/or an equalizer, i.e. frequency-dependentgain control.

The audio signals captured by the microphones M1, M2 of the microphonearrangement 42 are supplied as input to the beam former 58, and theoutput signal provided by the beam former 58 is supplied to themixing/adding unit 62. In addition, the audio signals of at least one ofthe microphones M1, M2 are supplied to the classification unit 60; inaddition, also the output of the beam former 58 may be supplied to theclassification unit 60. The classification unit 60 serves to analyze theaudio signals captured by the microphone arrangement 42 in order todetermine a present auditory scene category from a plurality of auditoryscene categories, i.e. the classification unit 60 serves to determinethe present acoustic environment. The output of the classification unit60 is supplied to the beam former 58, the mixing/adding unit 62 and theaudio signal processing unit 64 in order to control the audio signalprocessing in the central signal processing unit 28 by selecting thepresently applied audio signal processing scheme according to thepresent acoustic environment as determined by the classification unit60.

Also the audio signals captured by the external microphone 48 may besupplied to the classification unit 60 in order to be taken into accountin the auditory scene analysis.

The output of the audio input monitoring unit 66 may be supplied to theclassification unit 60, to the mixing/adding unit 62 and to the audiosignal processing unit 64 in order to select an audio signal processingscheme according to the presence of an external audio source 46 oraccording to the type of external microphone 48. For example, theexternal microphone 48 may be a boom microphone, one or a plurality ofomni-directional microphones or a beam-forming microphone. Depending onthe type of microphone, the audio input sensitivity and otherparameters, such as the choice between an energy-based VAD or a moresophisticated VAD based on direction of arrival in the classificationunit 60, may be adjusted automatically.

The audio signals captured by the auxiliary microphone 56 are suppliedto the mixing/adding unit 62 in order to be subtracted from the audiosignals captured by the microphone arrangement 42, for example, by usinga Wiener filter, in order to remove body noise and/or housing noise fromthe audio signals captured by the microphone arrangement 42.

The audio signals received at the audio input 44, 44′ are supplied tothe mixing/adding unit 62.

The output of the mixing/adding unit 62 is supplied to the audio signalprocessing unit 64.

The distance sensor 50 may comprise an acoustic, usually ultrasonic,and/or an optical, usually infrared, distance sensor in order to measurethe distance between the sound source, usually a speaking person towardswhich the microphone arrangement 42 is directed, and the microphonearrangement 42. To this end, the distance sensor 50 is arranged in sucha manner that it aims at the object to which the microphone arrangement42 is directed. The output of the distance sensor 50 is taken intoaccount in the audio signal processing unit 64 in order to select anaudio signal processing scheme according to the measured distance.

The acceleration sensor 52 serves to measure the acceleration acting onthe transmission unit 10—and hence on the microphone arrangement 42—inorder to estimate in which mode the transmission unit 10 is presentlyused. For example, if the measured acceleration is very low, it can beconcluded that the transmission unit 10 is used in a stationary mode,i.e. in a conference mode.

The orientation sensor 54 preferably is designed for measuring thespatial orientation of the transmission unit, and hence the microphonearrangement 42, so that it can be estimated whether the microphonearrangement 42 oriented essentially vertical or essentially horizontal.Such orientation information can be used for estimating the present usemode of the transmission unit 10. For example, an essentially verticalorientation is typical for a neck-worn/chest-worn mode.

By combining the information provided by the acceleration sensor 52 andthe orientation sensor 54 the best estimation of the present use mode isobtained. For example, an essentially horizontal position withoutsignificant acceleration is an indicator of a conference/restaurantmode, whereas an essentially horizontal position with acceleration ofsome extent is an indicator of a hand-held mode. In the hand-held mode,the distance measurement by the distance sensor 50 is most useful, sincein the hand-held mode the user may hold the transmission unit 10 in sucha manner that the microphone arrangement 52 points to a person speakingto the user. The orientation sensor 54 may comprise a gyroscope, a tiltsensor and/or a roll ball switch.

The output of the sensors 50, 52 and 54 is supplied to the audio signalprocessing unit 64 in order to select an audio signal processing schemeaccording to the measured values of the mechanical parameters of themicrophone arrangement 42 monitored by the sensors 50, 52 and 54. Inparticular, as already mentioned above, the information provided by thesensors 50, 52 and 54 can be used to estimate the present use mode ofthe transmission unit 10 in order to automatically optimize the audiosignal processing by selecting the audio signal processing scheme mostappropriate for the present use mode.

In the following, examples of such optimization of the audio signalprocessing are described by reference to FIGS. 3 to 5.

At the bottom in FIG. 3 an example of the gain as a function of theinput signal level (the corresponding dependency of the output signallevel on the input signal level is shown above in FIG. 3) of a defaultgain model is shown. In the example of FIG. 3 the gain is essentiallyconstant for medium input signal levels (from K1 to K2) while the gainis reduced for high input signal levels with increasing input signallevel (“compression”) and the gain is also reduced for low input signallevels (“soft squelch” or “expansion”).

FIG. 5 shows an example of the gain as a function of frequency of adefault gain model (curve A), which is relatively flat.

When the transmission unit 10 is hanging around the neck or is attachedto the chest of a person speaking to the user of the ear unit 12(“neck/chest mode”, which is indicated by an essentially verticalposition as measured by the orientation sensor 54), input levelsexceeding 75 dB-SPL can typically be expected for the speech signal tobe transmitted (this condition is indicated by the working point P1 inFIGS. 3 and 4). The compression reduces the gain in this case. In the“neck/chest mode”, input signals below a certain level, e.g. knee pointK2, can be considered to be mostly surrounding noise and/or clothingnoise and shall be compressed. Based on the information of the wearingmode, the release time of the compression algorithm can be increased toa few seconds, which avoids the background noise coming up in speechpauses.

A similar reduction of the overall gain may take place if the audioinput monitoring unit 66 detects that a chest microphone or a boommicrophone is connected to the transmission unit 10.

When the audio input monitoring unit 66 detects the presence of anexternal audio signal source 46, which typically is a music player, a“music mode” may be selected in which the dynamic range is increased,for example, by avoiding too strong compression in order to enhance thelistening comfort (an example is indicated in FIG. 4 by the curve M).

When the transmission unit 10 is in a horizontal position with virtuallyno movement, which is an indicator for the conference/restaurant mode inwhich the transmission unit 10 is placed on a table, the beam former 58should be switched to an omni-directional mode in which there is no beamforming, while the frequency-dependent gain should be optimized forspeech understanding. According to FIG. 5, speech understanding may beenhanced by reducing the gain at frequencies below and above the speechfrequency range, see curve C. Alternatively, the beam former 58 may beswitched to a zoom mode in which the direction of the beamformer isautomatically adjusted to the direction of the most intense soundsource.

As already mentioned above, an essentially horizontal position of thetransmission unit 10 with relative movements of some extent indicatesthat the transmission unit 10 is carried in the hand of the user of theear unit 12. In this case, a beamforming algorithm with enhanced gain atlower input levels (as indicated by the arrow in FIG. 4) would be thefirst choice. The gain applied at lower input levels may depend on themeasured distance to the sound source, with a larger distance requiringhigher gain. Such enhanced gain at lower input levels is indicated bythe curves H1 and H2 in FIG. 4. In addition, an enhanced roll-off at lowand high frequencies, i.e. at frequencies outside the speech frequencyrange, may be applied in order to emphasize speech signals while keepinglow frequency and high frequency noises at reduced gain levels, seecurves B and C of FIG. 5.

The information obtained by the distance sensor 50 with regard to thedistance of the microphone arrangement 42 to the sound source may beused to set the level-dependent and/or frequency-dependent gain and/orthe aperture angle of the beam former 58 according to the measureddistance.

While various embodiments in accordance with the present invention havebeen shown and described, it is understood that the invention is notlimited thereto, and is susceptible to numerous changes andmodifications as known to those skilled in the art. Therefore, thisinvention is not limited to the details shown and described herein, andincludes all such changes and modifications as encompassed by the scopeof the appended claims.

1. A method for providing hearing assistance to a user, comprising:capturing audio signals by an internal microphone arrangement andsupplying the captured audio signals to a central signal processingunit; estimating whether external microphone is connected to the centralsignal processing unit in order to supply external audio signals to thecentral signal processing unit and estimating the type of externalmicrophone by sensing at least one electrical parameter of the externalmicrophone, and selecting an audio signal processing scheme according tothe estimated type of external microphone; processing, by said centralsignal processing unit, the captured audio signals and the externalaudio signals according to the selected audio signal processing scheme;transmitting the processed audio signals to stimulating means worn at orin at least one of the user's ears via a wireless audio link; andstimulating the user's hearing by said stimulating means according tothe processed audio signals.
 2. The method of claim 1, wherein thewireless audio link is an FM link.
 3. The method of claim 1, wherein theexternal audio signal supply device is an external audio source.
 4. Themethod of claim 3, wherein an audio signal processing scheme accordingto a music mode is selected wherein a dynamic range is increased withregard to a default audio signal processing scheme.
 5. The method ofclaim 1, wherein the external audio signal supply device is a anexternal microphone, wherein a type of external microphone is estimatedby sensing at least one electrical parameter of the external microphone,and wherein the audio signal processing scheme is selected according tothe estimated type of external microphone.
 6. The method of claim 5,wherein an audio signal processing scheme is selected in which an audioinput sensitivity is adjusted according to the estimated type ofexternal microphone.
 7. The method of claim 5, wherein an audio signalprocessing scheme is selected in which a type of voice activity detectoris selected according to the estimated type of external microphone. 8.The method of claim 1, further comprising analyzing, by a classificationunit of the central signal processing unit, the audio signals in orderto determine a present auditory scene category from a plurality ofauditory scene categories, and selecting an audio signal processingscheme according to the determined present auditory scene category. 9.The method of claim 1, comprising: measuring at least one mechanicalparameter selected from the group consisting of an acceleration of theinternal microphone arrangement, a spatial orientation of the internalmicrophone arrangement and a distance of the internal microphonearrangement to a sound source; and selecting an audio signal processingscheme according to the measured at least one mechanical parameter. 10.The method of claim 9, wherein the internal microphone arrangementcomprises at least two spaced-apart microphones capable of acousticbeamforming.
 11. The method of claim 10, wherein, if an essentiallystationary horizontal orientation of the internal microphone arrangementis measured, an audio signal processing scheme corresponding to aconference mode is selected in which, with regard to a default audiosignal processing scheme, a frequency-dependent gain is optimized forspeech understanding.
 12. The method of claim 11, wherein an audiosignal processing scheme is selected in which there is no beamforming.13. The method of claim 11, wherein an audio signal processing schemeincluding an acoustic zoom mode is selected in which a direction of thebeamformer is automatically adjusted to a direction of the most intensesound source.
 14. The method of claim 10, wherein, if an essentiallyhorizontal non-stationary orientation of the microphone arrangement ismeasured, an audio signal processing scheme according to a hand-heldmode is selected in which beamforming takes place and wherein, withregard to a default audio signal processing scheme, a gain at low inputlevels is enhanced.
 15. The method of claim 14, wherein the enhancementof the gain at low input levels increases with increasing measureddistance of the microphone arrangement to the sound source.
 16. Themethod of claim 15, wherein an audio signal processing scheme isselected in which, with regard to a default audio signal processingscheme, a gain at frequencies below and above a speech frequency rangeis reduced in order to emphasize speech signals.
 17. The method of claim10, wherein, if an essentially vertical orientation of the microphonearrangement is measured, an audio signal processing scheme according toa neck/chest mode is selected in which, with regard to a default audiosignal processing scheme, at least one of an overall gain is reduced andthe release time is increased to more than one second in order to reducebackground noise.
 18. The method of claim 10, wherein an audio signalprocessing scheme is selected wherein at least one of a gain, which isat least one of level-dependent and frequency-dependent, and an apertureangle of the beamformer is selected according to a measured distance ofthe microphone arrangement to the sound source.
 19. A system forproviding hearing assistance to a user, comprising: a internalmicrophone arrangement for capturing audio signals; means for estimatingwhether an external microphone is connected to a central signalprocessing unit and for estimating the type of external microphone bysensing at least one electrical parameter of the external microphone,wherein the central signal processing unit is for processing thecaptured audio signals and the external audio signals supplied by theexternal microphone according to an audio signal processing schemeselected according to the estimated type of external microphone; andmeans for transmitting the processed audio signals via a wireless audiolink to means worn at or in at least one of the user's ears forstimulating a hearing of the user according to the processed audiosignals, said transmitting means comprising a transmitter portion and areceiver portion.
 20. The system of claim 19, wherein the transmittingmeans is adapted to establish a radio frequency audio link.
 21. Thesystem of claim 19, wherein the internal microphone arrangement, theestimating means, the central signal processing unit and the transmitterportion are integrated within a portable unit.
 22. The system of claim21, wherein the portable unit is a hand-held unit.
 23. The system ofclaim 21, wherein the portable unit is adapted to be worn around aperson's neck/on a person's chest.
 24. The system of claim 21, whereinthe external audio signal supply device is an external audio signalsource, and wherein the portable unit comprises an input for supplyingaudio signals from an external audio signal source to the audio signalprocessing unit.
 25. The system of claim 21, wherein the external audiosignal supply device is an external microphone, and wherein the portableunit comprises an input for supplying audio signals captured by theexternal microphone to the audio signal processing unit.
 26. The systemof claim 25, wherein the estimating means is for estimating a type ofexternal microphone connected to the input by sensing at least oneelectrical parameter of the external microphone, and wherein the centralsignal processing unit is adapted to select an audio signal processingscheme according to the estimated type of external microphone.
 27. Thesystem of claim 21, wherein the portable unit comprises means formeasuring at least one mechanical parameter selected from the groupconsisting of an acceleration of the internal microphone arrangement, aspatial orientation of the internal microphone arrangement and adistance of the internal microphone arrangement to a sound source, andwherein the central signal processing unit is for processing thecaptured audio signals according to an audio signal processing schemeselected according to the measured at least one mechanical parameter.28. The system of claim 27, wherein the measuring means comprises atleast one of an acoustic distance sensor and an optical distance sensor.29. The system of claim 27, wherein the measuring means comprises atleast one of a gyroscope, a tilt sensor and a roll ball switch.
 30. Thesystem of claim 27, wherein the measuring means are for measuring thespatial orientation of the microphone arrangement within a verticalplane.
 31. The system of claim 19, wherein the central signal processingunit comprises a classification unit for analyzing the audio signals inorder to determine a present auditory scene category from a plurality ofauditory scene categories, and wherein the central signal processingunit is adapted to select an audio signal processing scheme according tothe determined present auditory scene category.
 32. The system of claim31, wherein the classification unit comprises a voice activity detector.33. The system of claim 19, wherein the stimulating means is part of ahearing instrument comprising at least one microphone.
 34. The system ofclaim 33, wherein the receiver portion is integrated within or connectedto the hearing instrument.
 35. The system of claim 21, wherein theportable unit comprises an auxiliary microphone in close mechanical andacoustical contact with the housing of the portable unit for capturingaudio signals representative of at least one of body noise and housingnoise, and wherein the audio signal processing unit is adapted to usethe audio signals captured by the auxiliary microphone for removing atleast one of body noise and housing noise from the audio signalscaptured by the microphone arrangement.
 36. The system of claim 35,wherein the audio signal processing unit is adapted to use a Wienerfilter in order to subtract the audio signals captured by the auxiliarymicrophone from the audio signals captured by the microphonearrangement.